summaryrefslogtreecommitdiffstats
path: root/audio/filter/af_volume.c
Commit message (Collapse)AuthorAgeFilesLines
* audio: fix format function consistency issueswm42015-06-261-1/+1
| | | | | | | | | | | Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
* af: move af_from_dB() function to af_volume.cwm42015-06-231-3/+11
| | | | | And also simplify it (it certainly had the most awkward API you could think of for such a simple function).
* af_volume: add a replaygain fallback optionwm42015-06-231-0/+5
|
* audio: make softvol scale cubicwm42015-05-221-2/+5
| | | | | | | | This brings the volume control closer to what is percepted as linear volume change. Adjust the --softvol-max default to roughly the old maximum (roughly doubles the gain).
* Update license headersMarcin Kurczewski2015-04-131-5/+4
| | | | Signed-off-by: wm4 <wm4@nowhere>
* audio: make all format query shortcuts macrosKevin Mitchell2015-04-031-1/+1
| | | | | af_fmt_is_float and af_fmt_is_planar were previously inconsistent with AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
* af_volume: use refcounted frameswm42015-01-131-8/+15
|
* af_volume: dump applied replaygain in verbose modewm42015-01-041-1/+5
|
* af_volume: fix calculations including replay-gainMohammad Alsaleh2014-06-281-2/+2
| | | | | | | | | | | | | rgain is not an additive value. It's a multiplier/gain. Previous behaviour produced negative level values in some cases (when rgain < 1.0) which caused volume to be louder when its value was lowered. CC: @mpv-player/stable Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com> Signed-off-by: wm4 <wm4@nowhere>
* Add more constwm42014-06-111-1/+1
| | | | | | | While I'm not very fond of "const", it's important for declarations (it decides whether a symbol is emitted in a read-only or read/write section). Fix all these cases, so we have writeable global data only when we really need.
* af_volume: fix clang -Wsometimes-uninitializedKevin Mitchell2014-04-131-1/+1
|
* demux: move metadata-based replaygain decoding out of af_volumeAlessandro Ghedini2014-04-041-80/+9
|
* af_volume: use replaygain side dataAlessandro Ghedini2014-04-041-7/+19
|
* af_volume: fix replaygainwm42014-03-271-2/+3
| | | | | | | | This was accidentally broken in commit b72ba3f7. I somehow made the wild assumption that replaygain adjusted the volume relative to 0% instead of 100%. The detach suboption was similarly broken.
* af_volume: don't print missing replaygain tags as errorwm42014-03-141-1/+1
| | | | There's no reason to. Audio files often lack them.
* af_volume: add detach optionwm42014-03-141-0/+4
| | | | | | Maybe this should be default. On the other hand, this filter does something even if the volume is neutral: it clips samples against the allowed range, should the decoder or a previous filter output garbage.
* af_volume: separate softvol volume control from replaygain levelwm42014-03-141-5/+8
| | | | | | | | | Currently, both replaygain adjustment and user volume control (if softvol is enabled) share the same variable. Sharing the variable would cause especially if --volume is used; then the replaygain volume would always be overwritten. Now both gain values are simple added right before doing filtering.
* af_volume: remove double-negated suboptionwm42014-03-141-3/+3
| | | | | You had to use "no-replaygain-noclip" to set this option. Rename it, so that only one negation is needed.
* af_volume: add support for replaygain pre-amp and clipping preventionAlessandro Ghedini2014-03-131-11/+74
|
* af_volume: add replaygain supportAlessandro Ghedini2014-03-131-0/+22
| | | | | | | | | This adds the options replaygain-track and replaygain-album. If either is set, the replaygain track or album gain will be automatically read from the track metadata and the volume adjusted accordingly. This only supports reading REPLAYGAIN_(TRACK|ALBUM)_GAIN tags. Other formats like LAME's info header would probably require support from libav.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-1/+1
|
* audio/filter: change filter callback signaturewm42013-12-051-3/+3
| | | | | | | | | The new signature is actually closer to how it actually works, and someone who is not familiar to the API and how it works might make fewer fatal mistakes with the new signature than the old one. Pretty weird. Do this to sneak in a flags parameter, which will later be used to flush remaining data of at least vf_lavfi.
* audio/filter: remove unneeded AF_CONTROLs, convert to enumwm42013-11-181-2/+2
| | | | | | | | The AF control commands used an elaborate and unnecessary organization for the command constants. Get rid of all that and convert the definitions to a simple enum. Also remove the control commands that were not really needed, because they were not used outside of the filters that implemented them.
* audio: drop "_NE"/"ne" suffix from audio formatswm42013-11-151-5/+5
| | | | | | You get the native format by not appending any suffix to the format. This change includes user-facing names, e.g. for the --format option.
* audio/filter: fix mul/delay scale and valueswm42013-11-121-1/+0
| | | | | | | | | | | | | Before this commit, the af_instance->mul/delay values were in bytes. Using bytes is confusing for non-interleaved audio, so switch mul to samples, and delay to seconds. For delay, seconds are more intuitive than bytes or samples, because it's used for the latency calculation. We also might want to replace the delay mechanism with real PTS tracking inside the filter chain some time in the future, and PTS will also require time-adjustments to be done in seconds. For most filters, we just remove the redundant mul=1 initialization. (Setting this used to be required, but not anymore.)
* af: don't require filters to allocate af_instance->data, redo bufferswm42013-11-121-1/+0
| | | | | | | | | | | | | Allocate af_instance->data in generic code before filter initialization. Every filter needs af->data (since it contains the output configuration), so there's no reason why every filter should allocate and free it. Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min(). Interestingly, most code becomes simpler, because the new function takes the size in samples, and not in bytes. There are larger change in af_scaletempo.c and af_lavcac3enc.c, because these had copied and modified versions of the RESIZE_LOCAL_BUFFER macro/function.
* af_volume: add support for non-interleaved audiowm42013-11-121-16/+25
|
* audio/filter: prepare filter chain for non-interleaved audiowm42013-11-121-4/+5
| | | | | | | | | | | | | | | | | | Based on earlier work by Stefano Pigozzi. There are 2 changes: 1. Instead of mp_audio.audio, mp_audio.planes[0] must be used. 2. mp_audio.len used to contain the size of the audio in bytes. Now mp_audio.samples must be used. (Where 1 sample is the smallest unit of audio that covers all channels.) Also, some filters need changes to reject non-interleaved formats properly. Nothing uses the non-interleaved features yet, but this is needed so that things don't just break when doing so.
* af_volume: use only one volume setting for all channelswm42013-11-091-22/+14
| | | | | | | | In theory, af_volume could use separate volume levels for each channel. But this was never used anywhere. MPlayer implemented something similar before (svn r36498), but kept the old path for some reason.
* af: replace macros with too generic nameswm42013-10-261-2/+3
| | | | | | | | Defining names like min, max etc. in an often used header is not really a good idea. Somewhat similar to MPlayer svn commit 36491, but don't use libavutil, because that typically causes us sorrow.
* af_volume: some more cosmeticswm42013-10-261-27/+15
|
* af_volume: switch to new option parsingwm42013-10-261-31/+18
|
* af_volume: uncrustifywm42013-10-261-94/+85
| | | | Also, use more C99 and remove "register" keywords.
* af_volume: don't change volume if nothing is to be changedwm42013-10-261-0/+2
| | | | | On the float path. Note that this skips clipping, but we expect that everything on the audio-path is pre-clipped anyway.
* af_volume: remove unused featureswm42013-10-261-50/+1
| | | | | | Roughly follows MPlayer svn commits 36492 and 36493. We also remove the volume peak reporting. (There are much better libavfilter filters for this, I think.)
* audio/filter: remove useless af_info fieldswm42013-10-231-6/+4
| | | | | | | Drop the author and comment fields. They were completely unused - not even printed in verbose mode, just dead weight. Also use designated initializers and drop redundant flags.
* mixer, af_volume: use linear values instead of dBwm42013-09-191-4/+9
| | | | | | | Softvol always used a linear multiplier for volume control. This was converted to dB, and then back to linear in af_volume. Remove this non- sense. We still try to keep the command line argument to af_volume in dB, though.
* audio: add some setters for mp_audio, and require filters to use themwm42013-05-121-6/+3
| | | | | | | | | | | | | | | | mp_audio has some redundant fields. Setters like mp_audio_set_format() initialize these properly. Also move the mp_audio struct to a the file audio.c. We can remove a mysterious line of code from af.c: in.format |= af_bits2fmt(in.bps * 8); I'm not sure if this was ever actually needed, or if it was some kind of "make it work" quick-fix that works against the way things were supposed to work. All filters etc. now set the format correctly, so if there ever was a need for this code, it's definitely gone.
* audio/filter: remove unused AF_CONTROLswm42013-05-121-16/+0
| | | | Was unused, has never been used.
* audio: remove float processing optionwm42013-04-131-6/+2
| | | | | | | | | | | | | | The only thing this option did was changing the behavior of af_volume. The option decided what sample format af_volume would use, but only if the sample format was not already float. If the option was set, it would default to float, otherwise to S16. Remove use of the option and all associated code, and make af_volume always use float (unless a af_volume specific sub-option is set). Silence maximum value tracking. This message is printed when the filter is destroyed, and it's slightly annoying. Was enabled due to enabling float by default.
* Rename directories, move files (step 1 of 2) (does not compile)wm42012-11-121-0/+226
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-10 05:39:26 +0000